395 Commits

Author SHA1 Message Date
pkasting@chromium.org
005b6fffe6 Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails.
BUG=none
TEST=none
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39649004

Cr-Commit-Position: refs/heads/master@{#8222}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8222 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 19:42:17 +00:00
tommi@webrtc.org
4161715e3f Remove ChangeUniqueID.
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.

It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.

BUG=
R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37849004

Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 12:14:13 +00:00
glaznev@webrtc.org
8501ee632b Support VP8 HW decoding on devices with Exynos codec.
R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8160 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 23:07:19 +00:00
jiayl@webrtc.org
dacdd9403d Reland r7980:
Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908

BUG=4068, crbug/446908
R=juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 17:33:34 +00:00
kjellander@webrtc.org
a02d76845f Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness
Disabling the test on all platforms since it's likely it can happen
on any platform, even if it's only been observed on Win x64 Release.

Running tests in parallel is a huge performance benefit to the team,
since it approximately reduces build cycle with 60-75%.

BUG=4219
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8138 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 14:34:52 +00:00
tommi@webrtc.org
586f2eda0d Change GetStreamBySsrc to not copy StreamParams.
This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple.  Also, we can use lambdas now :)

BUG=
R=perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8131 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 23:00:41 +00:00
jiayl@webrtc.org
cceb166a3f Fix a use-after-free when sending queued messages is aborted for blocked channel.
BUG=4187
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8119 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 00:55:10 +00:00
tommi@webrtc.org
4fb7e25843 Update StatsReport and by extension StatsCollector to reduce data copying.
Summary of changes:
* We're now using an enum for types instead of strings which both eliminates unecessary string creations+copies and further restricts the type to a known set at compile time.
* IDs are now a separate type instead of a string, copying of Values is not possible and values are const to allow grabbing references outside of the statscollector.
* StatsReport member variables are no longer public.
* Consolidated code in StatsCollector (e.g. merged PrepareLocalReport and PrepareRemoteReport).
* Refactored methods that forced copies of string (e.g. ExtractValueFromReport).
* More asserts for thread correctness.
* Using std::list for the StatsSet instead of a set since order is not important and updates are more efficient in list<>.

BUG=2822
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8110 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 11:36:18 +00:00
braveyao@webrtc.org
fedb9ea6bc Correct the class name in peerconnection_jni.cc.
BUG=4194
TEST=Manual Test
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8106 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 07:57:06 +00:00
jlmiller@webrtc.org
5f93d0a140 Update libjingle license statements at top of talk files for consistency
BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 21:36:13 +00:00
tommi@webrtc.org
8e327c45d0 Update StatsCollector's interface in preparation of more changes.
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.

The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.

The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.

BUG=2822
R=perkj@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8095

Review URL: https://webrtc-codereview.appspot.com/36829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8097 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 20:41:26 +00:00
tommi@webrtc.org
43e54e36bf Revert 8095 "Update StatsCollector's interface in preparation of..."
> Update StatsCollector's interface in preparation of more changes.
> 
> This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
> 
> The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
> 
> The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
> 
> BUG=2822
> R=perkj@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/36829004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8096 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 17:34:23 +00:00
tommi@webrtc.org
5b76fd79df Update StatsCollector's interface in preparation of more changes.
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.

The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.

The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.

BUG=2822
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8095 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 16:49:33 +00:00
phoglund@webrtc.org
f9d3555ec3 Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test.
The was was really, really difficult to run before because you needed
a custom env with both LD_PRELOAD and library path. Now the script will
set up the correct library path and be more transparent about what it
requires.

BUG=None
TESTED=locally
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8093 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 13:57:59 +00:00
decurtis@webrtc.org
487a444215 Add stats collection for the data channel.
BUG=1805
R=bemasc@chromium.org, hta@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8083 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 22:55:07 +00:00
tkchin@webrtc.org
ef2a5dd398 Update AppRTCDemo UI.
- Removed log box. Debug logs still available through lldb.
- Remote video displayed in aspect fill format.
- Provide a hangup button.
- Added Default-568.png so we display properly on iPhone5+.

BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8081 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 22:38:21 +00:00
guoweis@webrtc.org
61c1247224 Fix a case where empty candidate id is used
BUG=4161
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8071 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 06:53:07 +00:00
pthatcher@webrtc.org
fd630a50d2 Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior.
R=decurtis@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8067 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 23:19:06 +00:00
kwiberg@webrtc.org
2ebfac5649 Remove COMPILE_ASSERT and use static_assert everywhere
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.

R=aluebs@webrtc.org, andrew@webrtc.org, hellner@chromium.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 10:51:54 +00:00
kwiberg@webrtc.org
3df38b442f Unify the two copies of compile_assert.h
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.

R=aluebs@webrtc.org, andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 11:37:48 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
glaznev@webrtc.org
be40eb0579 Allow 720x1280 frames encoding on Android.
Current maximum encoder width and height for Android is
hard-coded to 1280x720, so if device is rotated to portrait
orientation only part 720x1280 camera frame is extracted and
scaled to 1280x720. Increasing maximum height to 1280 allows
feeding video encoder with rotated 720x1280 frames directly
without scaling.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8042 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 17:55:47 +00:00
perkj@webrtc.org
81134d019d Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory.
In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is  specified in the call to CreatePeerConnectionFactory.

This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread.

Note that both Chrome and the Android implementation use an external signaling thread.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +00:00
pthatcher@webrtc.org
9657265f39 Revert "Accept incoming pings before remote answer is set to reduce connection latency."
This reverts r7980.

It was causing the ICE connected state to happen while still in the new state rather than going through the checking state, which was causing an ASSERT to fire, which was causing a crash.

Review URL: https://webrtc-codereview.appspot.com/41429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8031 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 19:08:27 +00:00
decurtis@webrtc.org
8af11042cb Avoid reading past end of string in GetLine.
BUG=3881
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8017 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 19:15:51 +00:00
tkchin@webrtc.org
4e5115ae73 RTCPeerConnectionFactory: Explicitly create new worker and signaling threads.
There should be no change in behavior, since this is what the default
constructor does.

BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8007 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 06:35:18 +00:00
glaznev@webrtc.org
f6a9714760 Remove peer connection and signaling calls from UI thread.
- Add separate looper threads for peer connection and websocket
signaling classes.
- To improve the connection speed start peer connection factory
initialization once EGL context is ready in parallel with the room
connection.
- Add asynchronious http request class and start using it in
webscoket signaling and room parameters extractor.
- Add helper looper based executor class.
- Port some of henrika changes from
https://webrtc-codereview.appspot.com/36629004/ to fix sensor
crashes on non L devices - will remove the change if CL will
be submitted soon.

R=jiayl@webrtc.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8006 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 22:24:09 +00:00
pbos@webrtc.org
9eacb8cc59 Make P2PTestConductor use VirtualSocketServer.
Permits running JsepPeerConnectionP2PTestClient in parallel.

TBR=juberti@webrtc.org
BUG=2598
TEST=third_party/gtest-parallel/gtest-parallel -w 128 -r 100 out/Debug/libjingle_peerconnection_unittest --gtest_filter=JsepPeerConnectionP2PTestClient.*

Review URL: https://webrtc-codereview.appspot.com/37459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7988 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 09:03:19 +00:00
jiayl@webrtc.org
c5fd66dcdf Accept incoming pings before remote answer is set to reduce connection latency.
BUG=4068
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7980 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 19:23:37 +00:00
tkchin@webrtc.org
7ce4a584aa Add initWithCoder to RTCEAGLVideoView.
Allows for proper OpenGL initialization if view is created from
storyboard.

BUG=3896
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7970 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 20:47:35 +00:00
stefan@webrtc.org
742386a136 Enable payload-based padding by default and remove the API.
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 15:33:17 +00:00
tommi@webrtc.org
209df9bf77 Change MockStatsObserver to grab values inside of OnComplete.
This is done since StatsReportCopyable is going away and the list of
supported properties of the mock class is known.
StatsReports holds a list of pointers to objects that cannot be cached,
so this is a simple way to grab the values when they're available.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 14:09:05 +00:00
guoweis@webrtc.org
950c518251 Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Committed: https://code.google.com/p/webrtc/source/detail?r=7906

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7925 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 23:01:31 +00:00
pthatcher@webrtc.org
e2b7585bc2 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository.
R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:09:08 +00:00
guoweis@webrtc.org
55360ae402 Revert "Add adapter_type into Candidate object."
This reverts commit aaf02cc2d4f696345ce0e6d5715f2cfa22aea689.

BUG=
TBR=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 05:28:10 +00:00
guoweis@webrtc.org
aaf02cc2d4 Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 23:03:10 +00:00
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00
tommi@webrtc.org
e2e199b894 Clean up StatsObserver's OnComplete methods (address TODOs).
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 13:22:54 +00:00
buildbot@webrtc.org
032b802a8c (Auto)update libjingle 82121498-> 82126219
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7896 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:48:07 +00:00
tommi@webrtc.org
dd0601fbcf Remove unneeded ctor and add a more practical one
The default constructor isn't necessary, so I'm removing it.
I'm adding another one so that we can (later) make |type| const.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7895 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:47:49 +00:00
tommi@webrtc.org
69bc5a300f Add thread asserts to StatsCollector.
Also adding a "ForTest" postfix to a test-only method.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:44:48 +00:00
pbos@webrtc.org
fb108b5a28 Revert r7885.
Breaks compile step of other code where network name of
cricket::Candidate is used.

TBR=guoweis@webrtc.org,juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/31229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 08:04:50 +00:00
pbos@webrtc.org
18a3896bd2 Revert r7886:7887.
Broke build steps in other code that uses securetunnelsessionclient.cc
and others.

TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/36439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:03:04 +00:00
pthatcher@webrtc.org
dee76f3b89 Move the obvious/easy Jingle-specific code into webrtc/libjingle.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 21:04:42 +00:00
guoweis@webrtc.org
8c9d79a29d Add adapter_type into Candidate object.
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 19:21:14 +00:00
tommi@webrtc.org
c57310b982 Switch kStatsValueName* constants to be enums instead of char*.
This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7884 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 17:41:28 +00:00
pthatcher@webrtc.org
40b276ea7b Cleanup little things found when refactoring.
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/33519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 02:44:30 +00:00
tommi@webrtc.org
aa2c342c10 Add back a constructor to fix FYI build.
TBR=perkj

Review URL: https://webrtc-codereview.appspot.com/24349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7854 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 20:23:06 +00:00
tommi@webrtc.org
c9d155faeb Move implementation of types in statstypes. to its cc file.
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7850 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 18:18:06 +00:00
henrika@webrtc.org
a954c07ee1 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 16:22:09 +00:00